I'm using DUNDi. My lookup returns 'IAX2' for the tech, and 'dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101' for the destination. How do I dial this? I've tried dialling it with: "Dial" "IAX2/dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101" passed from my AGI script, but the other endpoint (xxx.187.142.204) is returning: May 11 09:23:41 NOTICE[18846]: chan_iax2.c:6796 socket_read: Rejected connect attempt from xxx.187.142.203, who was trying to reach '3254101@' What should be after the '@' symbol? Why is it empty? The IAX dialling parameters are pretty confusing. Do I need to set anything special up in iax.conf on th e remote side, or does DUNDi handle all that? Thanks, Doug.
Did you set up a dundi iax user in iax.conf? On Thu, 11 May 2006, Douglas Garstang wrote:> I'm using DUNDi. > > My lookup returns 'IAX2' for the tech, and 'dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101' for the destination. > > How do I dial this? > > I've tried dialling it with: > > "Dial" "IAX2/dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101" > > passed from my AGI script, but the other endpoint (xxx.187.142.204) is returning: > > May 11 09:23:41 NOTICE[18846]: chan_iax2.c:6796 socket_read: Rejected connect attempt from xxx.187.142.203, who was trying to reach '3254101@' > > What should be after the '@' symbol? Why is it empty? The IAX dialling parameters are pretty confusing. Do I need to set anything special up in iax.conf on th e remote side, or does DUNDi handle all that? > > Thanks, > Doug. > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198
No... do you have an example of what that looks like? I get more matches on google for 'the early history of hungarian cabinet making' than I do for DUNDi examples.> -----Original Message----- > From: Aaron Daniel [mailto:amdtech@shsu.edu] > Sent: Thursday, May 11, 2006 9:43 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Dialling a DUNDi Route > > > Did you set up a dundi iax user in iax.conf? > > On Thu, 11 May 2006, Douglas Garstang wrote: > > > I'm using DUNDi. > > > > My lookup returns 'IAX2' for the tech, and > 'dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101' for > the destination. > > > > How do I dial this? > > > > I've tried dialling it with: > > > > "Dial" "IAX2/dundi:q9sgTFkVMBFdmp0IDX1bYQ@xxx.187.142.204/3254101" > > > > passed from my AGI script, but the other endpoint > (xxx.187.142.204) is returning: > > > > May 11 09:23:41 NOTICE[18846]: chan_iax2.c:6796 > socket_read: Rejected connect attempt from xxx.187.142.203, > who was trying to reach '3254101@' > > > > What should be after the '@' symbol? Why is it empty? The > IAX dialling parameters are pretty confusing. Do I need to > set anything special up in iax.conf on th e remote side, or > does DUNDi handle all that? > > > > Thanks, > > Doug. > > > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Aaron Daniel > Computer Systems Technician > Sam Houston State University > amdtech@shsu.edu > (936) 294-4198 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Thanks. I got it working. Now I've hit a HUGE snag. We're doing all of our call routing from a database accessed from AGI. When we trunk calls from one asterisk system over to another via IAX to terminate the call, the dialling parameters are defined by what's in the dial command on the second system, not the first. This is a big problem. :(> -----Original Message----- > From: Florian Overkamp [mailto:florian@obsimref.com] > Sent: Thursday, May 11, 2006 10:09 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Dialling a DUNDi Route > > > Douglas Garstang wrote: > > No... do you have an example of what that looks like? I get more > > matches on google for 'the early history of hungarian > cabinet making' > > than I do for DUNDi examples. > > > [dundi] > type=user > dbsecret=dundi/secret > context=dundi-e164-local > > > > Best regards, > Florian > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
> Douglas Garstang wrote: > > We're doing all of our call routing from a database accessed from > > AGI. When we trunk calls from one asterisk system over to > another via > > IAX to terminate the call, the dialling parameters are defined by > > what's in the dial command on the second system, not the first. This > > is a big problem. :( > > Errrrh, ok, I have a very faint idea of what you are saying. But what > are you trying to achieve ?What am I trying to achieve? Uhm... a carrier grade, highly redundant (ie multiple servers), VOIP solution with advanced business(not residential) features such as findme/followme, incoming and outgoing blacklisting/whitelisting(user/org/company level), user/prefix defined pic codes and rate centers, intra company 4 digit extension dialling, feature codes, user defined internal, external, override caller id and on and on - all provisionable and maintainable via a web interface (don't forgot the multiple servers!)!.... does that answer your question? When you IAX trunk a call from Asterisk A to Asterisk B, you can't pass the ring time and ring options of the original SIP call between servers. Doug.
We are using a backend MySQL database for call flow, not user agent registration info. Just how, exactly, is a backend database going to replicate registration data between Asterisk servers? Realtime has been documented NOT to work with multiple Asterisk systems. If you like I can dig up the list messages from Kevin Fleming on this subject. Realtime also has way too many limitations. Doug -----Original Message----- From: Florian Overkamp [mailto:florian@obsimref.com] Sent: Thu 5/11/2006 3:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Dialling a DUNDi Route Douglas Garstang wrote: > What am I trying to achieve? Uhm... a carrier grade, highly redundant > (ie multiple servers), VOIP solution with advanced business(not > residential) features such as findme/followme, incoming and outgoing > blacklisting/whitelisting(user/org/company level), user/prefix > defined pic codes and rate centers, intra company 4 digit extension > dialling, feature codes, user defined internal, external, override > caller id and on and on - all provisionable and maintainable via a > web interface (don't forgot the multiple servers!)!.... does that > answer your question? Yaddayaddah. Don't go at this lighthearted. You can use DUNDi for call distribution between asterisk nodes and automatic discovery. However, depending on how big your site(s) will be, it may be worthwhile to take a look at database integration (i.e. the realtime API in asterisk). It will in most cases give you a finer level of granularity than DUNDi will. > When you IAX trunk a call from Asterisk A to Asterisk B, you can't > pass the ring time and ring options of the original SIP call between > servers. Correct, same applies for using 'switch' in your dialplan. Once the call is gone, it's gone. DUNDi was not designed for that type of applications. Maybe you are better served with a good dynamic database on multiple servers :-) My EUR 0,02 F. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 5738 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060511/2d9934d5/attachment.bin
Patrick, Dug all day... found nothing! -----Original Message----- From: Patrick [mailto:asterisk@puzzled.xs4all.nl] Sent: Thu 5/11/2006 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Dialling a DUNDi Route On Thu, 2006-05-11 at 10:33 -0600, Douglas Garstang wrote: [snip] > When you IAX trunk a call from Asterisk A to Asterisk B, you can't pass the ring time and ring options of the original SIP call between servers. Iirc you can pass variables on the IAX link to the other side. Maybe you can use those settings to define the ringtime etc. Don't recall how to pass them though so you need to do some digging there. Regards, Patrick _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
I'm not sure if you have considered this, but if you were using SIP between the Asterisk servers you can definitely achieve this using X-headers. Regards, - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas Garstang Sent: Thursday, May 11, 2006 11:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dialling a DUNDi Route Patrick, Dug all day... found nothing! -----Original Message----- From: Patrick [mailto:asterisk@puzzled.xs4all.nl] Sent: Thu 5/11/2006 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] Dialling a DUNDi Route On Thu, 2006-05-11 at 10:33 -0600, Douglas Garstang wrote: [snip] > When you IAX trunk a call from Asterisk A to Asterisk B, you can't pass the ring time and ring options of the original SIP call between servers. Iirc you can pass variables on the IAX link to the other side. Maybe you can use those settings to define the ringtime etc. Don't recall how to pass them though so you need to do some digging there. Regards, Patrick _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.
> -----Original Message----- > From: Leif Madsen [mailto:asterisk.leif.madsen@gmail.com] > Sent: Friday, May 12, 2006 12:27 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Dialling a DUNDi Route > > > On 5/12/06, Florian Overkamp <florian@obsimref.com> wrote: > > Douglas Garstang wrote: > > > We are using a backend MySQL database for call flow, not > user agent > > > registration info. Just how, exactly, is a backend > database going to > > > replicate registration data between Asterisk servers? Realtime has > > > been documented NOT to work with multiple Asterisk systems. If you > > > like I can dig up the list messages from Kevin Fleming on this > > > subject. Realtime also has way too many limitations. > > > > You're thinking inside the box. I'm not saying Kevin is > wrong. You can > > probably design a database that uses a per-asterisk set of > tables and > > uses triggers or a stand alone daemon to manually replicate the data > > between machines. If realtime doesn't fit your need, consider > > automatically generating extensions.conf etc. from databases using > > scripts and templates. > > Use func_odbc to get information from your database into the dialplan > -- then you don't need to pass that information along through the path > via DUNDi, you just look it up as you need it, then use it. > > At least that's what I'm doing and it works great. Tilghman > Lesher is my hero :)We tried something similar with the MySQL dialplan command. It didn't work. To support findme/followme, we needed to nest database lookups, and the MySQL dialplan command wasn't able to remember the state of queries as they nested.